speaker

一个完整的基于Matlab的DTW模型算法及高效算法程序，能快速识别数字0-9,运行testdtw即可。(An excellent MATLAB program for the algorithm as a DTW model. It can recognise the number of 0-9. Try it, just run testdtw.)

关于如何使用fpga实现语音识别 最新版本(err)

块和子带自适应滤波器matlab源文件，内有详细说明及运行结果图(Block and Subband Adaptive Filters)

voice recognition using mfcc

是信号处理的基础，完整的基于HMM的语音识别系统，外文资料里面的源代码。( Is the basis of the signal processing, Complete HMM-based speech recognition system, Foreign materials inside the source code.)

主要是研究光纤通信系统中的克服PMD效应的电域均衡技术，构建电域均衡器的模型并进行仿真实验。找到补偿PMD效应的较理想的电域均衡器结构。(Is to study, fiber-optic communication systems to overcome the effects of PMD electrical domain equalization technology, building a power domain equalizer model and simulation experiments. PMD compensation effect to find a better balance of power domain structure.)

谱减法的程序 希望可以被大家使用～～ 做的比较一般但运行起来还是可以的(Spectral subtraction procedure can be hoped that the use of ~ ~ U.S. to do the more general but can be up and running the)

Chapter 2 Low Bit-Rate Speech Coding In this chapter an overview is given of speech coding techniques at several bit rates. Most of them use Linear Prediction. This overview is not meant to be complete its purpose is to make the reader somewhat familiar with Linear Predictive Coding which is necessary for a proper understanding of later chapters. Section 2.1 treats the subject of quantisation and coding. In section 2.2 a description of speech production and speech sounds is given. Coders based on linear prediction can be considered as being based on a simple speech production model. This model is explained in section 2.3. Section 2.4 describes various speech coding algorithms and techniques. Section 2.5 briefly describes some measures for the quality of coded speech.(In this chapter an overview is given of speech coding techniques at several bit rates. Most of them use Linear Prediction. This overview is not meant to be complete its purpose is to make the reader somewhat familiar with Linear Predictive Coding which is necessary for a proper understanding of later chapters. Section 2.1 treats the subject of quantisation and coding. In section 2.2 a description of speech production and speech sounds is given. Coders based on linear prediction can be considered as being based on a simple speech production model. This model is explained in section 2.3. Section 2.4 describes various speech coding algorithms and techniques. Section 2.5 briefly describes some measures for the quality of coded speech.)

这是基于隐马尔可夫模型的连续语音识别代码，不同于dtw,这是个完整的工程，我把需要的voicebox也放进去了，一共用到9个子函数，包括模板的训练算法何识别算法，是完全可以使用的。(This is based on the hidden Markov model of continuous voice recognition code, different the dtw, which is a complete project, I need to put the voicebox into a total of nine sub-functions, including the template training algorithm to identify the algorithm, Is fully available.)

语音压缩和回声消除，支持LINUX和其他平台(voip echo cancellation)